similar to: ASR & ACD

Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source & destination? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/48076c6b/attachment.htm

Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk

Hi Everyone, I am in search of a reliable open source tool for the real time monitoring of ASR & ACD, so any help or suggestions will be highly appreciated. Regards, Faisal Rehman -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111020/d283a5af/attachment-0001.htm>

Hi, What is good monitoring software to run to get the above info and more? Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080913/21a3679a/attachment.htm

Dear fellows, how are you? I?m offering TDM routes for Brazil (landline and mobile destinations) with low prices, TDM ccts (no GSM), ASR and ACD great. Pre paid, by paypal. If you have interest, please just let me know. With Best Regards Josue -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was

Hello, All of the Opus quality studies that I've seen focused on human-perceived quality. I'm interested to know of any experience with machined "perceived" quality, particularly related to speech recognition or biometrics. I'm also interested in folks thoughts on optimizing Opus for ASR. For example, removing certain classes of comfort noise, filtering non-speech bands,

Hello Luca, Thank you for your response. I?m familiar with speech recognition and TTS, but new to MRCP. Yes, the 100k options is used for names in a directory listing. In the pre-MRCP support, Nuance ASR used API events/methods for the application to tell ASR when the prompt was playing and when it stopped. If ASR detected speech, it would signal an event so we would stop playing the prompt.

Is there is an open source ACD component for Asterisk? Likewise, is there an open source CTI component that will work with Asterisk? Regards, Jim O'Brien -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/e6ee98da/attachment.htm

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

Hello, I'm in the early stages of designing an Emergency calling service IVR application. The IVR application asks simple one or two questions like "which is the postal code of the area you are currently calling from ?" "Is the correct ?". The expected values are a 5-digits number like "twenty-five-thousand-two-hundreds-twelve" or

Good Day again list, Encountered another problem in the ACD queue... If I use the ADDQueueMember to dynamically add members as foolows, exten => 403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM}) lets assume I called extension 403 from my extension 2204. then a caller (extension 2203) enters into the techsupport queue I am able to receive the support call on my phone (extension 2204

Hellow, i need the following requirements with asterisk : 1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? 2) How call barging can set up in asterisk ? 3) How call recording can set up in asterisk? Thanks mahboob System Engr SSL -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a couple of real quick questions for ya'll. I am trying to setup * in a test scenario to emulate my company's 20 agent ACD. 1) First, I am curious if anyone has any configs that they can throw up on the wiki or just send via e-mail. I have a bit of it pieced together and am able to make some very basic queued calls, but I am sure that there are much smarter people out there

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk??

Hi all Arnaud, Some newly purchased Ellipse ASR-model UPS find their way to my office for a first evaluation test. It concerns the Ellipse 1500 and Ellipse 1000 models. As you know we are still working with hidups. Peripheral recognition, start of NUT, mains short interruptions management, all that work fine... as expected ;-) But changes occured on the delayed shutdown procedure after the

Can anyone point me in the right direction of info regarding ACD methods available in Asterisk. As far as I can see there are time based ring strategies available but I cannot find any info regarding skills based routing or queue priorities. Also do the current time based ring strategies work globally. What I mean by this is if an agent is a member of more than one queue then would the ACD

I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong.

Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is